CELP model uses an extensive codebook of excitation waveforms, and a
closed loop search mechanism for identifying the best excitation in
the set for every frame of input speech, without rigidly classifying
the input as voiced or unvoiced. The above adaptivety leads to natural
sounding speech at relatively low bit rates.Further, if the spectral
envelope in the speech model is estimated in a backward manner, based
on a recent past history of quantized speech, rather than a current
part of unquantized speech, the result is the low delay CELP system of
Figure 
.
The LD-CELP algorithm has been shown to provide high-quality coding of
telephone speech at 16 kbit/s, which turns out to be 2 bits per sample
with 8 kHz sampling. With 16 kHz sampling of 7 kHz speech, and a
target of 32 kbit/s, the bit rate is again 2 bits per sample. Wideband
speech is however, harder to code since the data is highly
unstructured at high frequencies and the spectral dynamic range is
very high.
Figure: (a) Forward (b) Backward adaptive system for CELP